Configuring WebRTC in Asterisk (FreePBX)
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The telephony server must be accessible from the Internet, i.e. it must have a static IP.
Must have a working SSL/TLS certificate.
Asterisk version not less than 11 (), but for convenience preferably 15 () or at least 13 ().
When using browser-based softphone, wss (WebSocket Secure) must be configured on the Asterisk server, and the port must be open to the outside (usually 8089) ().
This is configured in the admin interface:
Advanced Settings > Asterisk Builtin mini-HTTP section > HTTPS Bind Port
Port range for RTP (typically 10000-20000) ().
It is configured in the administrator interface:
Settings → Asterisk SIP Settings → General SIP Settings Tab
This example describes how to configure WebRTC in an already running FreePBX server:
is available at <example>.com and has an SSL/TLS certificate;
FreePBX version 16, Asterisk version 15;
wss is configured on port 8089;
all other relevant ports are open.
If the "Change To CHAN_PjSIP Driver" button (see below, in the internal number setting) is available, you do not need to do anything in this section.
Settings ➝ Advanced Settings
SIP Channel Driver
both
Settings ➝ Asterisk SIP Settings
Tab “SIP Legacy Settings [chan_sip]”
Verify that the variables are set:
Other SIP Settings
icesupport = yes
media_encryption = dtls
tlscipher = AES256-SHA
Go to the FreePBX configuration interface and log in: https://<example>.com/
Go to section Applications ➝ Extensions
Button “+ Add Extension” ➝ “+ Add New SIP (Legacy) [chan_sip] Extension” (other versions of FreePBX may be “Add New Chan_SIP extension”)
In the tab General:
User Extension
1001
Display Name
1001
Press the button Submit
After that, open editing of the newly created extension in the list of extensions (in the line with 5001 in the Actions column, click the edit button with a pencil icon).
Go to the tab Advanced
First change the SIP Driver to PjSIP:
Enable WebRTC defaults
Yes
Media Encryption
DTLS-SRTP (not recommended)
Press the button Submit
Then change the SIP Driver back:
Change SIP Driver
Change To CHAN_SIP Driver
Now to make the remaining customizations:
Transport
All - WSS Primary
Enable AVPF
Yes
Force AVP
Yes
Enable ICE Support
Yes
Enable rtcp Mux
Yes
Enable Encryption
Yes (SRTP only)
Section DTLS
Enable DTLS
Yes (there should be)
Use Certificate
<example>.com
Press the button Submit
Press the button Apply Config
Let's use the sipML5 live demo from Doubango as an example:
First, the "Expert mode?" button.
The advanced settings page should open in the next tab
WebSocket Server URL
wss://.com:8089/ws
Press the button Save
Back to the main tab
Private Identity
1001
Public Identity
sip:1001@<example>.com
Password
Copy from FreePBX settings
Extension: 1001 ➝ General tab ➝ Edit Extension ➝ Secret
Realm
<example>.com
Button Login
The word Connected should appear at the top above the Registration heading.
Now you can make the call.
Under the Call control heading, dial the desired number (*43 for an echo call or a cell phone number with 8 first).
Press the button Call ➝ Audio
There's a call that should go out.
Profit!
Initial support for WebRTC in Asterisk starting with version 11:
New chan_pjsip driver in Asterisk 12:
PjSIP is available by default in Asterisk 15: